10 Oct
2014
10 Oct
'14
5:42 a.m.
Hello,
you have to provide the sip trace taken on kamailio server, capturing
the traffic on both sides. What you provided is missing any description
and ips of sender and receiver.
As a guess, if you don't get bye on kamailio, very likely you didn't do
record_route() for invite.
Cheers,
Daniel
On 25/09/14 12:22, balu wrote:
Hi
I am using kamailio with rtp proxy module. I have 2 questions /issues .
1. When caller or callee ends the call the other end call is not
disocnnecting .
UA is pjsip based and behind NAT router. Present call flow is
pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP
proxy----> ThridParty SIP Server
UA local ip : 192.168.2.11
UA public IP : 89.78.92.23
Kamailio Public ip: 94.50.203.32
Third party Sip server : 76.42.89.25
Here When I disconnect call from either side , it is not
disconnecting other side .
2. My second requirement is , how can I define port of third party
server .
for example if have 3 or 4 sip servers with different sip registration
ports other tahn 5060
How can I route registration requests coming from UAs to different
ports of third party servers.
Please bear my ignorance I am new to kamailio .Hope some experts will
help me here .
Attached kamailio config and SIP trace taken from kamailio server
Thank you
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Daniel-Constantin Mierla
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